CN110753295A

CN110753295A - Calibration method for customizable personal sound delivery system

Info

Publication number: CN110753295A
Application number: CN201811545860.8A
Authority: CN
Inventors: 克里斯托弗·阿诺德·杰弗里; 詹姆斯·亚历山大·菲尔丁; 亚历克斯·约翰·阿夫利克
Original assignee: Aldra Pte Ltd
Current assignee: Aldra Pte Ltd
Priority date: 2018-07-23
Filing date: 2018-12-14
Publication date: 2020-02-04
Anticipated expiration: 2038-12-14
Also published as: EP3827598A4; EP3827598A1; AU2019312034A1; WO2020019020A1; CN110753295B

Abstract

A calibration method for a customizable personal sound delivery system is provided. A method for calibrating a sound delivery system includes: transmitting a sequence of command codes from a remote user interface device for a sound delivery system specifying predetermined characteristics of a test sound; receiving a sequence of command codes at a communication component of a sound delivery system; providing a sequence of command codes to a processing component of the sound delivery system; reproducing a predetermined test sound by the selected at least one audio transducer under control of the at least one processor in accordance with a command code sequence; measuring a characteristic of a test sound reproduced by the sound delivery system with a reference SPL meter proximate to the audio transducer; comparing the measured characteristic of the reproduced sound with a predetermined characteristic of the test sound; generating a mapping of the specified test sound to sound reproduced by the at least one audio transducer; and storing the mapping in an electronic memory associated with the remote user interface device.

Description

Calibration method for customizable personal sound delivery system

RELATED APPLICATIONS

This application is related to australian innovation patent No. 2016100861a4 entitled "a custom publication specific delivery system" in the name of the applicant entitled "a patent patented at 7/2016 and is incorporated herein by reference.

Technical Field

The invention relates to a calibration method for a sound transmission system of the following type and to a sound transmission system subjected to the calibration method of the invention: sound delivery systems of the type described involve audio transducers, such as headphones, ear buds or in some cases bone conduction transducers, and may be customized by the user to take into account the user's auditory response.

Background

Any reference to methods, apparatus or documents of the prior art should not be taken as an acknowledgment or any form of evidence or admission that they form or form part of the common general knowledge in australia or any other country.

Different people have different auditory responses. For example, young people are often able to hear audio frequencies at higher frequencies than older people. As people age, or suffer from hearing impairment due to exposure to loud sounds, their hearing tends to decline and their auditory response to the frequency range changes.

It is known to provide programmable hearing aids that can be set up by testing the user by an audiologist to compensate for the user's reduced hearing acuity or partial hearing loss. However, such systems require the user to make an appointment to perform the hearing test, often using expensive and cumbersome test equipment, and requiring the hearing aid to be set up by a technician.

It has further been recognized that, in order to effectively compensate for hearing loss, it is important to precisely calibrate the audio transducer disposed in the hearing compensation system during manufacture and/or assembly so that tests performed by the user provide consistent and reliable performance, including regardless of the interface device provided by the user.

It is an object of the present invention to provide a method for calibrating a sound delivery system for automated hearing tests. It is another object of the invention to provide a customizable personal sound delivery system that is pre-calibrated using this method for ease of use, and that can automatically test a user's hearing and then adjust its sound delivery parameters based on the test results.

Disclosure of Invention

According to a first aspect of the present invention there is provided a method for calibrating a sound delivery system having: a processing component; a data communication component coupled to the processing component; and at least one audio transducer mounted with the at least one processor of the processing assembly and responsive to the at least one processor to transmit sound to a user, the method comprising the steps of:

transmitting a sequence of command codes from a remote user interface device for a sound delivery system specifying predetermined characteristics of a test sound;

receiving a sequence of command codes at a communication component of a sound delivery system;

providing a sequence of command codes to the processing component of a sound delivery system;

reproducing a predetermined test sound by the selected at least one audio transducer under control of the at least one processor in accordance with a command code sequence;

measuring a characteristic of a test sound reproduced by the sound delivery system with a reference meter proximate the audio transducer;

comparing the measured characteristics of the reproduced sound with predetermined characteristics of the test sound;

generating a mapping of a specified test sound to sound reproduced by the at least one audio transducer; and

the mapping is stored in an electronic memory associated with the remote user interface device.

Preferably, the transmitting step involves using wireless transmission using a local or near field communication standard such as Wi-Fi or bluetooth.

The user interface device suitably comprises a portable computing device, for example, a smart watch, a smart phone, a tablet or a laptop computer.

Preferably, the test sounds or tones comprise a sequence of discrete sounds having different frequencies and different Sound Pressure Levels (SPLs) within each frequency, suitably covering a typical range of human hearing.

Preferably, the test sound is in the frequency range from 10Hz to 30kHz, suitably in the frequency range from 20Hz to 20kHz, most preferably including 100Hz, 250Hz, 500Hz, 1kHz, 2kHz, 4kHz, 8kHz and 16kHz, and has a Sound Pressure Level (SPL) within each discrete sound frequency of from-10 dB to 120dB, suitably from 0dB to 110 dB.

Desirably, each of the discrete sounds in the sequence is of equal duration and, suitably, is separated from adjacent sounds by a period of silence. Suitably, the duration of the sound is in the range from 0.1 milliseconds to 5 seconds, suitably in the range from 100 milliseconds to 1 second, and the intervening silent period is in the range from 0.1 milliseconds to 5 seconds, suitably in the range from 100 milliseconds to 1 second.

Suitably, the storing step involves storing the test sound map in a code library utilized by an audio application interface of the sound delivery system. Ideally, the sound delivery system comprises non-volatile electronic memory arranged to store a code library. More preferably, the code library is stored remotely in a database and associated with an interface application of the sound delivery system for downloading on request by means of the interface application.

The sound delivery system may be an audiological testing device, such as a hearing aid, a set of headphones, or other head-mountable hearing device containing an audio transducer.

In another form, there is also provided a sound delivery system comprising: a processing component comprising at least one processor and electronic memory; a user interface coupled to the at least one processor; at least one audio transducer responsive to the processing component to transmit sound to a user; and an electronic memory accessible by the at least one processor and storing: instructions for a processor to: determining, for a user, a compensation weight at each of a plurality of audio frequencies based on a user response to sound transmitted via an audio transducer via the interface and transmitting audio signals modified according to the determined weights to the user via the audio transducer; a code library utilized by an audio application interface of the sound delivery system; wherein the sound transmitted via the transducers used to determine the compensation weights is generated by a transducer processor mounted within a transducer portion comprising at least one audio transducer; and wherein the sound delivery system is calibrated according to the method set out above.

Preferably, the processing assembly is mounted with at least one audio transducer; suitably, the at least one audio transducer is in the form of a set of headphones comprising a pair of speakers.

In a second aspect of the present invention, there is provided a sound transmission system comprising: at least one processing component; an interface coupled to at least one processing component; and at least one audio transducer responsive to the at least one processing component to transmit sound to a user; wherein the at least one processing component is arranged to determine a compensatory weight at each of a plurality of audio frequencies for the user based on a user response to sound transmitted via the audio transducer via the interface, and to transmit an audio signal modified according to the determined weights to the user via the audio transducer.

In a preferred embodiment of the invention, the sound delivery system comprises an interface part comprising the user interface and a transducer part comprising at least one audio transducer, wherein the first interface part and the transducer part comprise respective data communication components for data communication therebetween.

Preferably, the at least one processing assembly comprises: at least one interface processor mounted within the interface portion and coupled to the user interface; and at least one transducer processor mounted within the transducer portion and arranged to process sound signals into sound for transmission by the audio transducer.

Preferably, the data communication component is arranged for wireless data communication. For example, the data communication component may be arranged to enable data communication according to the bluetooth standard.

In a preferred embodiment of the invention the interface part comprises a smartphone, but it may alternatively be a tablet, laptop or desktop computer, for example.

According to another aspect of the present invention, there is provided a sound transmission system comprising: at least one processing component; an interface coupled to at least one processing component; at least one audio transducer responsive to the at least one processing component to transmit sound to a user; and electronic memory accessible by the at least one processing component and storing: instructions for the processor to determine, for a user, a backoff weight at each of a plurality of audio frequencies based on a user response to sound transmitted via an audio transducer via the interface and transmit an audio signal modified according to the determined weights to the user via the audio transducer.

In still another aspect of the present invention, there is provided an automatic audiological testing apparatus including: a processing component having at least one processor; an electronic memory in communication with the processing component and containing instructions for execution by the at least one processor; a user interface in communication with the processing component; and at least one audio transducer mounted with the processing assembly and responsive to the at least one processor to transmit sound to a user; wherein the electronic memory stores instructions for the processor to: determining, for a user, a backoff weight at each of a plurality of audio frequencies based on user responses to sound at a plurality of different frequencies via the interface, wherein sound transmitted via a transducer used to determine the backoff weight is generated by a transducer processor installed within a transducer portion comprising at least one audio transducer; and wherein the audiological testing device is calibrated according to the method set out above.

According to yet another aspect of the present invention, there is provided a set of headphones comprising: left and right loudspeakers for conveying sound to a user; at least one processor configured to receive a gain adjustment weight for a user for each of a plurality of predetermined frequencies; wherein the processor is arranged to convert the audio signal to the frequency domain, to apply the gain adjustment weights to the audio signal in the frequency domain and to convert the adjusted audio signal back to the time domain for conveying the adjusted audio signal to a user via the loudspeaker.

According to yet another aspect of the present invention, there is provided a method for transmitting sound to a user, comprising: presenting sounds and cues of different frequencies to a user to determine an audiological model of the user comprising a set of gain adjustment weights for each of the different frequencies; and adjusting the audio signal according to the adjustment weight to thereby deliver the adjusted audio signal to the user to compensate for the hearing deficiency of the user.

Preferably, the method comprises facilitating adjustment of the weights by the user to introduce a user-selected frequency equalization parameter for each of the plurality of frequency bands.

It will therefore be appreciated that in one embodiment of the present invention, a sound delivery system is provided that includes a processing assembly, wherein a user interface is coupled to the processing assembly. At least one audio transducer is provided to transmit sound to a user in response to the processing component. Typically, the audio transducer is a loudspeaker of a pair of headphones or earphones, but it may also be a bone conduction transducer. The at least one processing component is arranged to determine a compensatory weight at each of a plurality of audio frequencies for the user based on a user response to sound transmitted via the audio transducer via the interface and to transmit an audio signal modified according to the determined weights to the user via the audio transducer.

Additional features and advantages of the present invention are described in, and will be apparent from, the detailed description of the presently preferred embodiments and the accompanying drawings.

Drawings

Preferred features, embodiments and variations of the present invention will be apparent from the following detailed description, which provides sufficient information for a person skilled in the art to carry out the invention. The detailed description is not to be taken in any way as limiting the scope of the preceding inventive content of this invention. The detailed description will refer to the several figures as follows:

figure 1 is a high level diagram of a sound delivery system according to a preferred embodiment of the first aspect of the present invention in use;

FIG. 2 is a block diagram of the electronic circuitry of the transducer portion of the sound delivery system;

fig. 3A to 3D are first parts of schematic circuit diagrams substantially corresponding to the block diagram of fig. 2;

FIGS. 4A-4B are second portions of the circuit schematic of FIG. 2;

FIG. 5 is a high level block diagram of a user interface portion of the voice delivery system in the form of a smart phone;

6-10 are screen shots of screens presented to a user via a smartphone;

FIG. 11 is a block diagram illustrating a modeling method according to an embodiment of the invention;

FIGS. 12 and 13 are screen shots of screens presented to a user via a smartphone;

fig. 14 includes three frequency domain spectrums. On the left side is the hearing response spectrum of a person with normal hearing to the test audio signal. In the middle is the hearing response spectrum of a person with a reduced hearing response in the high frequency band to the test audio signal. On the right side is the perceived audio response to the test signal after it has been gain adjusted to compensate for the high frequency loss.

FIG. 15 is a flowchart of the steps performed by a sound delivery system delivering audio to a user;

FIG. 16 is a schematic diagram illustrating an apparatus employed in a calibration method of a further aspect of the present invention;

fig. 17 is a table showing an example of results obtained from the calibration method of the embodiment;

FIG. 18 is a flow chart of steps in a method for performing a calibration method using the components shown in FIG. 16 to produce the results tabulated in FIG. 17.

Detailed Description

Referring now to fig. 1, a sound delivery system 1 in use and applied to a user 3 is described. In the presently described preferred embodiment, the sound delivery system 1 comprises two main parts. The first part comprises a smartphone 5 or other computing device, e.g. a laptop, desktop or tablet computer. The smartphone 5 is in data communication with a second part of the sound transmission system, which is a transducer part, which in this embodiment comprises a headset 7, but which may equally be a set of ear buds or some other sound transmission device. In the presently described embodiment, the data communication between the smartphone 5 and the headset 7 is done wirelessly via bluetooth, but it may of course be established in other ways, e.g. via a wired connection or via other wireless protocols.

Fig. 2 is a high-level block diagram of the electronic circuitry contained within the headset 7. The circuit comprises a communication port 9 in the form of a bluetooth port for communicating with the smartphone 5. A processor in the form of a field programmable gate array 11 is coupled to the bluetooth port 9. As will be explained, the FPGA11 is configured to apply "weights", i.e. gain adjustment parameters, for different frequencies to the audio signal received from the smartphone through data uploaded from the smartphone 5. The output side of the FPGA11 is coupled to a digital-to-analog converter (DAC) 13. The DAC converts the digital audio signals from the FPGA into left and right stereo analog signals which are applied to

output amplifiers

19a, 19b via preamplifier 15 through

noise cancellation modules

17a, 17 b. The

output amplifiers

19a, 19b drive the electrical signals to the

vibration transducers

21a, 21 b. Typically, the

transducers

21a, 21b are loudspeakers, but they may alternatively be bone conduction transducers.

Fig. 3A to 3D are first parts of a circuit schematic corresponding to the block diagram 2 showing the bluetooth port 9, the FPGA11 and the DAC 13. Fig. 3A to 3D also show

programmable flash elements

23 and 25, the

programmable flash elements

23 and 25 being used to configure the FPGA, for example to set frequency gain adjustment weights that the FPGA will apply to an audio signal in use. The FPGA11 is a Cyclone IV EP4CE40F integrated circuit manufactured by Altera corporation and is configured to perform a fast fourier transform on audio signals received via the bluetooth port, apply gain weights in the frequency domain and then perform an inverse fast fourier transform to convert the digital signals back to the time domain.

Fig. 3A to 3D also show a clock module 27 and a power supply chip 29 for applying power to the respective components. All of these components are readily commercially available.

Fig. 4A-4B are second portions of circuit schematics (the first portions are depicted in fig. 3A-3D). Fig. 4A-4B illustrate a component level integrated circuit 15 that implements a left and right channel preamplifier 15. It also shows a component level integrated circuit 19 implementing left and

right output amplifiers

19a and 19 b.

The

output transducers

21a, 21b shown in the block diagram of fig. 2 comprise loudspeakers. However, as shown in fig. 4A-4B, they may instead include bone conduction transducers 31a, 31B. In this case, a splitter chip (demux chip)33 is provided to switch the output signal from the power amplifier 19 between the microphone and the bone conduction transducer as required. The splitter chip 33 is controlled by the FPGA via the splitter interface 32 on the chip.

Turning now to fig. 5, a block diagram of the smartphone 5 is depicted. The smartphone 5 comprises a plurality of modules capable of exchanging data and commands via a data bus 47. The various modules include:

a processor 35;

an electronic memory 37;

a communication module in the form of a bluetooth port 43 for communicating with a corresponding bluetooth port 9 of the transducer portion 7;

a telecommunications module 45 that allows the smartphone 5 to establish voice and data communications with a telecommunications network; and

a touch screen drive module 39 which drives the touch screen 41 and processes user data input received via the touch screen and passes it to the processor 35.

It will be appreciated that the architecture shown in fig. 5 is highly simplified and omits many of the elements found in smartphones. However, this will be sufficient for a person skilled in the art to understand the preferred embodiment of the invention.

The memory 37 stores instructions comprising a customising application, namely an "App" 38 which is executed, in use, by the processor 35 in order to perform a method in accordance with a preferred embodiment of an aspect of the present invention as will now be described. Once it is understood that the method will be apparent from the discussion below, the programming of App38 is simple and referring again to fig. 1, in use the user 3 puts on the headset 7 and turns on the headset and smartphone 5 so that bluetooth communication is established between them.

Then, the user operates the smartphone to start executing App38, for example, by clicking an icon of App displayed on the touch screen 41. The user 3 is then presented with the launch screen 49 shown in figure 6 and shortly thereafter with the control menu screen 51 shown in figure 7.

The control menu screen 51 presents the user 3 with 3

configuration options

51a, 51b, 51 c. The first option is my headphones 51 a. If the user has not used the App before and wishes to upload some equalizer style adjustments quickly, he/she may select the "my headphones" option 51 a. In response to this selection, processor 35 presents equalizer screen 58 shown in fig. 12. App is programmed so that user 3 can quickly set and upload equalizer preferences.

Alternatively, the user may select the second option "test history" option 51b if the user 3 has used the App38 before and has saved the hearing profile in advance. In response to selecting the "test history" option 51b, the processor causes a list view of the previous models to be displayed from which the user can select and upload to the headset 7 with or without equalizer coverage.

Finally, if user 3 is using APP38 for the first time, the user may select the "My Profile" option 51 c. Selecting the "my profile" option 51c causes the processor to invoke the audio modeling program and set up the personalized model to upload to the headset 7 with or without equalizer coverage. If equalizer coverage is applied, the gain adjustment weights that have been determined based on audiological tests are changed to account for the equalization preferences of the user. For example, if the user prefers a bass (bass) sound, the weight corresponding to the low frequency band is increased.

Upon selection of the My Profile option 51c, the processor 35 causes the screen to display prompts to prompt the user to help optimize the acoustic model as displayed by prompt screen 53 (FIG. 8) and prompt screen 55 (FIG. 9).

Once the user 3 operates the touch screen 41 to indicate that he/she is ready to run the audio model program, the app38 displays the interface screen 57 and directs the user 3 to respond to the software by pressing the "left" 57a or "right" 57b buttons. In doing so, the processor communicates with the headset 7 via a bluetooth link to cause the microphone in the headset to present beeps in the user's left or right ear, respectively.

The App then presents the screen to step the user through the modeling method 59 shown in fig. 11 and in accordance with a preferred embodiment of the present invention. Modeling method 59 includes the step of identifying a minimum perceptual headphone specific decibel threshold for the user at each of a number of frequency assessment points. The determined specific decibel threshold for the user is then saved in the audio model. The dB Threshold, which is variable for each frequency, is set in a box entitled "Stop Threshold", where the evaluation is stopped and the calculated dB value is saved. Alternatively, in the box "Stop not threshold", when the user has had severe hearing loss at that frequency and has maximized the hardware's capability, the program is stopped and the correction dB is set to the maximum value. In the flow chart, the label is "2 for 3 times at this level? "includes a portion of the error correction loop. In practice, the dB level will be presented to the user 3 times before the program exits. A user is considered to be able to hear it if they can hear it 2 out of 3. This is for the purpose of avoiding input errors and the like. The audio model is a set of parameters for the user, including decibel thresholds stored in digital memory 37.

Upon completion of method 59 for each of the frequency assessment points, the App successfully models the way in which the user perceives sound through headset 7. App38 then converts the perceived model into the graphical depiction 60 shown in fig. 13 for viewing. The graph in fig. 13 shows the hearing responses of the left and right ears of the user evaluated at each of a plurality of frequencies.

Referring to the spectrogram in fig. 14, method 59 finds a user's perceived gain deficiency 63 in each particular frequency band in the audio spectrum. Method 59 then calculates weights (i.e., gain correction factors) by which to gain adjust the future audio waveform in the frequency domain to correct the user's waveform back to the desired perceptual ensemble. These weighting coefficients are then uploaded from the smartphone 5 into the headset FPGA 7. The FPGA then uses the uploaded weights for dynamic real-time processing of uncompensated audio from the smartphone at runtime.

Then, App38 displays the equalizer screen 58 (fig. 12) to the user 3. Here the user has the option to either keep the overall correction according to the audiological model or to use such a basic level correction with equalizer coverage to allow additional personalization.

Once the user completes the customization and selects "upload," the model with or without equalizer coverage according to each user preference is translated into frequency-based coefficient corrections and uploaded to the pair of headphones for configuring the on-board signal processing corrections.

Fig. 15 is a flow chart showing how the FPGA is programmed to process the WAV file (or other audio file) using the determined gain adjustment weights, i.e., the audiological model, to thereby apply the weights generated in the audiological evaluation with or without equalizer coverage.

From the left to the right in fig. 15, the digital audio signal 61 is sent to the headset 7 via the bluetooth link 63. This received signal is sent to an on-board FPGA (item 11 of figure 2) for signal processing. Inside the FPGA, the time domain audio signal 61 is converted to the frequency domain 65 by using the FFT, and then gain adjusted for each corresponding frequency bin for the patient's personal correction weights 67 to produce a gain adjusted frequency representation of the signal. The FPGA11 then performs an IFFT to render the signal into the user-specific time-domain digital audio waveform 71. The digital audio waveform is then processed through a DAC (item 13 of fig. 2) and analog amplifier and possibly a noise cancellation module to drive the transducer of the headset.

It will therefore be appreciated that in a preferred embodiment of the present invention a sound delivery system 1 (fig. 1) is provided. The sound delivery system 1 includes at least one processing component, which in the presently described embodiment includes a smartphone processor 35 and a headset FPGA 11. The user interface is provided in the form of a smartphone touch screen 41 and a touch screen display driver unit 39. The touchscreen display driver unit 39 is coupled to the processor via a data bus 47. The sound transmission system 1 further comprises at least one audio transducer. For example, either or both of the

microphones

21a, 21b (fig. 4) and the

bone conduction transducers

31a, 31b (fig. 4) are provided. The bone conduction transducer responds to the signal from the FPGA via a suitable digital to analog converter and analog amplifier.

The at least one processing component comprises a processor 35 of the smartphone 5. The processor, as it executes instructions stored in digital memory 37, including App38, is arranged to determine, for the user, a back-off weight at each of a plurality of audio frequencies. The processor determines the weights based on a user response to sound transmitted via an audio transducer (e.g., a loudspeaker of the headphones 7) via an interface (e.g., the touch screen 41). The at least one processor further comprises an FPGA11, the FPGA11 being configured with the determined weights and thus capable of communicating to a user an audio signal resulting from modifying the audio signal according to the determined weights.

In the presently described embodiments of the invention, the user interface portion and the transducer portion of the sound delivery system are physically separate, but in data communication via a bluetooth connection. It will be appreciated that in other embodiments of the invention, the separation of the two units may not be so. For example, the headset may have a user interface mounted to the side, such as one or more buttons, coupled to the internal processor, such that the user may initiate an automated audiological assessment and then press one or other of the buttons to indicate an auditory threshold of the presented audio signal. Such an arrangement would not require the user interface portion and the transducer (i.e. earpiece) portion to be separate. In such embodiments, the processing component may comprise a single suitably programmed high frequency processor capable of not only running the audiological assessment method but also performing FFT and IFFT functions using gain adjustments according to the determined weights for the user.

Turning to fig. 16, a schematic diagram of an embodiment of a calibration device employed for factory calibration of a sound delivery system, here in the form of a customizable sound delivery system (or "SDS"), as described above, is shown. The SDS of this embodiment comprises a set of headphones 7 having a pair of

audio transducers

21a, 21b and a remote computing device, here in the form of a laptop or tablet 6. Note that the laptop and tablet have many components, such as a touchscreen 41', that are at least functionally identical to the smartphone 5 described above. The headset comprises a processor 11 and associated memory 12 which communicates with a remote device, such as a laptop computer 6, via a communication module 9. The laptop computer 6 may also communicate with remote storage accessible via a network (not shown), whether a public or virtual private network, such as a database 82 (sometimes referred to as "cloud storage") maintained in a remote storage facility.

The equipment necessary for calibration of the personal headset 7 includes a reference SPL meter 70, which reference SPL meter 70 is attached to a selected acoustic transducer, here the left speaker 21a, by an acoustic coupler 72 in order to exclude external noise during the calibration test. Suitable reference SPL meters include: digitech QM1592 professional sound level meter supplied by Jaycar Electronics in Australia, or bipolar ear station supplied by hong Kong miniDSP, particularly for headset kits (see alsowww.minidsp.com). It will be appreciated that it is not always economically feasible to calibrate each earpiece produced. Instead, a particular design or "model" of headphones may be manufactured to a quality standard of, for example, +/-2dB A, and from a given perspectiveThe produced representative headset was run through the calibration procedure described in relation to this embodiment. As an example, if the transducer pair is re-formulated or re-designed, a new calibration of the model variant will be performed. It will be appreciated that in other embodiments, for some specific medical applications, it may be desirable to calibrate each earpiece individually to achieve higher accuracy.

Fig. 18 is a top-level flow chart of a sequence of steps in an embodiment of the calibration method 100 of the second aspect of the present invention, here employing the apparatus and optional infrastructure shown in fig. 16. In step 102, a headphone body or earpiece 7 containing a representative SDS for a first audio transducer in the form of a speaker 21a is coupled to a reference sound pressure meter 70 through an acoustic coupler 72, directing the generated sound to a microphone 74 of the SPL meter. Then in step 104 the first of the sequence of command codes targeting the loudspeaker 21a is sent to the headphone assembly 7 requesting a discrete test tone of a specific frequency and sound pressure level to be output, e.g. requesting a command code of 100Hz at 0 dB. In step 106, the command code is confirmed through the communication interface 9 of the headphone assembly 7. In step 108, the processor 11 operates in response to the command code to cause the loudspeaker 21a to reproduce a test tone, which in turn is measured in step 110 by the reference level meter 70.

In step 112, the SPL readings obtained by the reference SPL meter 70 are recorded for transmission to a database associated with the user interface application of the SDS 1. Ideally, a mapping of the command code input to the processor 11 and the SPL readings obtained from the transducer 21a via the microphone 74 of the reference SPL meter 70 is constructed to produce a mapping table in the database. The SPL map resulting from the calibration may be stored, at least temporarily, in a database maintained locally in local memory 12 associated with processor 11, in the memory of the handheld device 6 controlling the calibration, or most desirably and ultimately in a remote database 82 maintained in cloud storage facility 80. The remote database 82 also suitably contains interface applications that are selectively downloaded to any compatible user interface device, and contains SPL maps for the particular model and/or production run of the headset 7 that has been calibrated. This effectively provides a single point of calibration, eliminating the need for "paired" interface devices and transducer hardware, which typically increases cost and/or does not facilitate achieving similar levels of accuracy.

In step 114, control passes back to step 104, where after a delay of 0.5s, command codes for subsequent test tones having the same frequency but different SPL levels (e.g., 10dB) are generated in step 104. The return loop 124 is then repeated with each desired SPL level (e.g., in 10dB steps to 100 dB).

At the end of the desired SPL level range, control drops from the next SPL in decision block 114 to decision block 116, where a subsequent frequency step is selected, e.g., 250Hz is restarted at a SPL level of 0 dB. Control then passes back to loop 124 and the 250Hz steps through each of the desired SPL levels.

At the end of each of the desired frequencies (and SPL levels within each frequency), e.g., 500Hz, 1kHz, 2kHz, 4kHz, 8kHz, and 16kHz, represented by loop 126, control drops to decision block 118, where, in step 102, the user will be prompted to move (if necessary) or switch (in the case of a bilateral meter) the acoustic coupler and reference SPL meter 70 to another one of the acoustic transducers, e.g., speaker 21 b. Control then returns to step 104 to repeat the test tone process for another transducer at each selected frequency and SPL level.

In use, as an example, if a command code "025-50" issued by the interface device 6 that requires a SPL of 50dB at 1kHz is reproduced as a test tone of 45dB by the left transducer 21a, the mapping associated with the interface application appropriate for the headphone model will be appropriately adjusted by the processor 11 during 1kHz tone generation. See the example results table depicted in fig. 17, where the mapping may be derived by an offset or difference between the requested and measured SPL results for the left transducer 21 a. It will be appreciated that for the right transducer 21b, a similar table portion will be generated over the entire range of desired frequencies and SPL.

In compliance with the statute, the invention has been described in language more or less specific as to structural features or methodological steps. The terms "comprise" and variations thereof such as "comprises" and "consisting of … … are used throughout in an inclusive sense rather than to exclude any additional feature. It is to be understood that the invention is not limited to the specific features shown or described, since the description herein of preferred forms includes presently preferred modes of carrying out the invention. The invention is, therefore, claimed in any of its forms or modifications within the proper scope of the appended claims appropriately interpreted by those skilled in the art.

Unless the context requires otherwise, throughout the description and claims (if present), the term "substantially" or "about" will be understood to be a value that is not limited to the range defined by the term.

Any embodiments of the present invention are meant to be illustrative only and not limiting. It is therefore to be understood that various other adaptations and modifications may be made to any of the described embodiments without departing from the spirit and scope of the invention.

Claims

1. A method for calibrating a sound delivery system having: a processing component; a data communication component coupled to the processing component; and at least one audio transducer mounted with the processing assembly and responsive to the processing assembly for delivering sound to a user, the method comprising the steps of:

transmitting a sequence of command codes from a remote user interface device for the sound delivery system specifying a predetermined characteristic of a test sound;

receiving the sequence of command codes at the communication component of the sound delivery system;

providing the sequence of command codes to at least one processor of the processing component of the sound delivery system;

reproducing a predetermined test sound by the selected at least one audio transducer under control of the at least one processor in accordance with the sequence of command codes;

measuring a characteristic of a test sound reproduced by the sound delivery system with a reference meter proximate to the audio transducer;

comparing the measured characteristic of the reproduced sound with the predetermined characteristic of the test sound;

storing the mapping in an electronic memory associated with the processing component.

2. The calibration method of claim 1, wherein the transmitting step involves using wireless transmission using a local or near field communication standard.

3. A calibration method according to claim 1 or 2, wherein the test sound comprises a sequence of discrete sounds having different frequencies and different sound pressure levels within each frequency, suitably covering a typical range of human hearing.

4. A calibration method according to claim 3, wherein the frequency of the test sound is in a frequency range from 10Hz to 30kHz, suitably from 20Hz to 20 kHz.

5. A calibration method according to claim 3, wherein the sound pressure level of the test sound is in the range from-10 dB to 120dB, suitably from 0dB to 110dB, per discrete sound frequency.

6. The calibration method of claim 3, wherein each of the discrete test sounds in the sequence has an equal duration and is separated from adjacent sounds by a period of silence.

7. The calibration method according to claim 6, wherein the discrete sound duration is in the range from 0.1 milliseconds to 5 seconds, suitably in the range from 100 milliseconds to 1 second.

8. The calibration method according to claim 6, wherein the silent period is in the range from 0.1 ms to 5 seconds, suitably in the range from 100 ms to 1 second.

9. The calibration method according to any one of claims 1 to 8, wherein the storing step involves storing the test sound map in a code library utilized by an audio application interface of the sound delivery system.

10. The calibration method of claim 9, wherein the code library is stored in a non-volatile portion of the electronic memory.

11. The calibration method according to claim 9 or 10, wherein the code library is also stored remotely in a database and associated with an interface application of the sound delivery system for downloading on request with the interface application.

12. A sound delivery system comprising:

a processing component comprising at least one processor and electronic memory;

an interface for a user coupled to the at least one processor;

at least one audio transducer responsive to the processing component to transmit sound to the user; and

wherein the electronic memory is accessible by the at least one processor and stores:

instructions for the processor to: determining a backoff weight at each of a plurality of audio frequencies for the user based on a user response to sound transmitted via the audio transducer via the interface and transmitting an audio signal modified according to the determined weights to the user via the audio transducer;

a code library utilized by an audio application interface of the sound delivery system;

wherein the sound transmitted via the transducers used to determine the compensation weights is generated by a transducer processor mounted within a transducer portion comprising the at least one audio transducer; and is

Wherein the sound delivery system is calibrated according to the method of any one of claims 1 to 11.

13. The sound delivery system of claim 12 wherein the processor of the processing assembly is mounted with the at least one audio transducer.

14. The sound delivery system of claim 13 wherein the audio transducer comprises a pair of speakers mounted in a set of headphones.

15. An automated audiological testing device comprising:

a processing component having at least one processor;

an electronic memory in communication with the processor and containing instructions for execution by the at least one processor;

a user interface in communication with the processor; and

at least one audio transducer mounted with the processing component and responsive to the at least one processor to transmit sound to a user;

wherein the electronic memory stores instructions for the processor to: determining a backoff weight for the user at each of a plurality of audio frequencies based on user responses to sound at a plurality of different frequencies via the interface;

Wherein the audiological testing device is calibrated according to the method of any one of claims 1 to 11.